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2. Reference Configuration
In the test configuration shown below, The Avaya S8720 Servers with Avaya G650 Media
Gateway are configured as an Access Element and support the Avaya 6408D+ Digital
Telephone, Avaya 9630 IP Telephone (H.323), and Avaya One-X Communicator (H.323). The
Avaya S8510 Server with Avaya G450 Media Gateway is configured as a Feature Server and
supports all of the Avaya SIP telephones shown. The Avaya 1100-Series IP Deskphone models
tested were the 1120E (4 line monochrome), 1140E (6 line monochrome), and the 1165E (8 line
color). Communication between the Communication Managers and Avaya Modular Messaging
is via Session Manager. Modular Messaging supports all telephones for voice messaging
coverage.
These application notes assume that Communication Manager and Session Manager are already
installed and basic configuration steps have been performed. Only steps relevant to SIP
telephone calling features will be described in this document. For further details on configuration
steps not covered in this document, consult the appropriate document in Section 10.
Equipment Software/Firmware
S8720 Server with G650 Media Gateway Avaya Aura™ Communication Manager 5.2.1,
S8510 Server with G450 Media Gateway Load 16.4, Update 17774
Avaya Aura™ Session Manager 5.2 Service
Pack 2, Load 5.2.2.0.522007
S8510 Server
Avaya Aura™ System Manager 5.2 Service
Pack 2, Load 5.2.2.0.522007
Avaya 9630 IP Telephone (SIP) 2.5
Avaya 9630 IP Telephone (H.323) 3.1
Avaya One-X Communicator (H.323) 5.2
Avaya 6408D+ Digital Telephone -
Modular Messaging Storage Server 5.2, Build 5.2-11.0
Modular Messaging Application Server 5.2, Build 9.2.150.13 (Patch 520008)
Avaya 1120E and 1140E IP Deskphones 03.00.56.01
Avaya 1165E IP Deskphone 03.02.05.20
These steps are performed from the Communication Manager System Access Terminal (SAT)
interface. Avaya and third party SIP telephones are configured as Off-PBX Stations (OPS) in
Communication Manager. Communication Manager does not directly control an OPS endpoint,
but its features and calling privileges can be applied to it by associating a local extension with
the OPS endpoint. Similarly, a SIP telephone in Session Manager is associated with an extension
on Communication Manager. SIP telephones register with Session Manager and use
Communication Manager for call origination and termination services, including Feature Name
Extension (FNE) support. Enter the save translation command after completing this section.
Step Description
1. Enter the display system-parameters customer-options command. Verify that there are
sufficient Maximum Off-PBX Telephones – OPS licenses. If not, contact an authorized
Avaya account representative to obtain additional licenses.
display system-parameters customer-options Page 1 of 10
OPTIONAL FEATURES
USED
Platform Maximum Ports: 44000 200
Maximum Stations: 450 60
Maximum XMOBILE Stations: 0 0
Maximum Off-PBX Telephones - EC500: 10 0
Maximum Off-PBX Telephones - OPS: 200 55
Maximum Off-PBX Telephones - PBFMC: 0 0
Maximum Off-PBX Telephones - PVFMC: 0 0
Maximum Off-PBX Telephones - SCCAN: 0 0
Note: Each SIP call between two SIP endpoints requires four SIP trunks for the duration
of the call. The license file installed on the system controls the maximum permitted.
display system-parameters customer-options Page 2 of 10
OPTIONAL FEATURES
Step Description
1. Enter the change ip-codec-set n command, where n is a number between 1 and 7,
inclusive. IP codec sets are used in Section 5.3 for configuring an IP network region to
specify which codec sets may be used within and between network regions. For the
compliance testing, G.722-64K, G.711MU, and G.729AB were used and Media
Encryption was set to none. If only one codec should be used, then only specify the one
that is to be used. Note that for G.729 interoperability between Avaya 1100-Series IP
Deskphones, Avaya 9600 Series SIP Telephones, and Avaya one-X Communicator in
Road-Warrior mode, the G.729A codec should be used, and the configuration file settings
for the 9600 SIP Telephone should include the line: SET ENABLE_G729 “1”.
change ip-codec-set 7 Page 1 of 2
IP Codec Set
Codec Set: 7
Media Encryption
1: none
2:
3:
Step Description
1. Enter the change ip-network-region n command, where n is a number between 1 and
250 inclusive and configure the following as shown in the display screen below:
Authoritative Domain – Set to avaya.com in this example. This should match the
SIP Domain value configured in Session Manager.
Intra-region IP-IP Direct Audio – Set to yes to allow direct IP-to-IP audio
connectivity between endpoints registered to Communication Manager or Session
Manager in the same IP network region.
Inter-region IP-IP Direct Audio – Set to yes to allow direct IP-to-IP audio
connectivity between endpoints registered to Communication Manager or Session
Manager in different IP network regions.
Codec Set – Set the codec set number as provisioned in Section 5.2.
display ip-network-region 1 Page 1 of 19
IP NETWORK REGION
Region: 1
Location: 1 Authoritative Domain: avaya.com
Name: Company X
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 7 Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048 IP Audio Hairpinning? y
UDP Port Max: 65535
DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y
Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS
Audio PHB Value: 46 Use Default Server Parameters? y
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6
Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
Step Description
1. Use the change node-names ip command to add a new node name for Session Manager.
change node-names ip Page 1 of 2
IP NODE NAMES
Name IP Address
SM1 10.1.2.170
default 0.0.0.0
procr 10.1.2.160
Step Description
1. Enter the command add signaling-group n, where n is an available signaling group and
configure the following as shown in the display screen below:
Group Type – Set to sip.
Transport Method – Set to tls.
IMS Enabled – Set to y.
Near-end Node Name - Set to procr.
Near-end Listen Port - Defaults to 5061 for TLS.
Far-end Node Name - Set to the node name configured in Section 5.4.
Far-end Listen Port - Defaults to 5061 for TLS.
Far-end Network Region - Set to the Region configured in Section 5.3.
Far-end Domain - Set to avaya.com in this example. This should match the SIP
Domain value configured in Session Manager.
Direct IP-IP Audio Connection – Set to y.
display signaling-group 60
SIGNALING GROUP
Step Description
1. Issue the command add trunk-group n, where n is an unallocated trunk group and
configure the following as shown in the display screen below:
Group Type – Set to the Group Type field to sip.
Group Name – Enter any descriptive name.
TAC (Trunk Access Code) – Set to any available trunk access code.
Signaling Group – Set to the Group Number field value configured in Section
5.5. (i.e., 60)
Number of Members – Allowed values are between 0 and 255. Set to a value
large enough to accommodate the number of SIP telephone extensions being used.
Note: Each SIP call between two SIP endpoints (whether internal or external) requires
two SIP trunk members for the duration of the call. The license file installed on the system
controls the maximum permitted.
display trunk-group 60 Page 1 of 21
TRUNK GROUP
Signaling Group: 60
Number of Members: 20
Step Description
1. Use the change system-parameters features command to administer system wide
features for the SIP telephones. Those related to features listed in Table 2 are shown in
bold. These are all standard Communication Manager features.
change system-parameters features Page 17 of 18
FEATURE-RELATED SYSTEM PARAMETERS
WHISPER PAGE
Whisper Page Tone Given To: paged
IP PARAMETERS
Step Description
1. Use the change dialplan analysis command to define the dial plan formats used in the
system. This includes all telephone extensions, Feature Name Extensions (FNEs), and
Feature Access Codes (FACs). To define the FNEs for the features listed in Table 2, a
Feature Access Code (FAC) must also be specified for the corresponding feature. In the
sample configuration, telephone extensions are five digits long and begin with 3, FNEs
are five digits beginning with 7, and the FACs have formats as indicated with Call Type
“fac”. Note that a FAC of “8” was used for AAR routing by a voice mail hunt group, the
configuration for which is not included in these Application Notes. See Reference [10]
for more information.
change dialplan analysis Page 1 of 12
DIAL PLAN ANALYSIS TABLE
Location: all Percent Full: 0
Step Description
1. Use the change class-of-service command to set the appropriate service permissions to
support the corresponding features (shown in bold). For the example, COS 1 was used.
On Page 2, set the value of VIP Caller to “y” only if all calls made by telephones with
this COS should be priority calls. Priority call indication (e.g., distinctive ring and display
of “Priority”) is only supported on Avaya Digital and 9600 Series IP telephones.
change cos Page 1 of 2
CLASS OF SERVICE
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
Auto Callback n y y n y n y n y n y n y n y n
Call Fwd-All Calls n y n y y n n y y n n y y n n n
Data Privacy n n n y n y y y y n n n n y y y
Priority Calling n y n n n n n n n y y y y y y n
Console Permissions y y y n n n n n n n n n n n n n
Off-hook Alert n n n n n n n n n n n n n n n n
Client Room n n n n n n n n n n n n n n n n
Restrict Call Fwd-Off Net n n y y y y y y y y y y y y y y
Call Forwarding Busy/DA n y n n n n n n n n n n n n n n
Personal Station Access (PSA) n n n n n n n n n n n n n n n n
Extended Forwarding All n y n n n n n n n n n n n n n n
Extended Forwarding B/DA n y n n n n n n n n n n n n n n
Trk-to-Trk Transfer Override n y n n n n n n n n n n n n n n
QSIG Call Offer Originations n n n n n n n n n n n n n n n n
Contact Closure Activation n n n n n n n n n n n n n n n n
0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15
VIP Caller n n n n n n n n n n n n n n n n
COR Number: 1
COR Description: Stations
FRL: 0 APLT? y
Can Be Service Observed? y Calling Party Restriction: none
Can Be A Service Observer? y Called Party Restriction: none
Partitioned Group Number: 1 Forced Entry of Account Codes? n
Priority Queuing? n Direct Agent Calling? y
Restriction Override: none Facility Access Trunk Test? n
Restricted Call List? n Can Change Coverage? n
1. Configure the station and off-PBX-station forms for each user in Communication
Manager. Then configure the corresponding user in Session Manager, being sure to
check the “Use Existing Stations” box (see Section 6).
2. Configure the user in Session Manager, being sure to leave the “Use Existing Stations”
box unchecked (see Section 6. Session Manager will automatically generate the
corresponding station and off-PBX-station information in Communication Manager.
Then use the change station command in Communication Manager to add other
configuration data, such as Coverage Path, MWI Served User Type, and additional call
appearances, if needed.
Method 2 was used in the sample configuration. For method 1, perform the following steps for
each user; then follow the steps in Section 6. For method 2, follow the steps in Section 6 first;
then use change station n to modify any station parameters as described below using the station
form in this section as a guide.
3. Proceed to Page 4 of the form and add the desired number of call-appr entries in the
BUTTON ASSIGNMENTS section. This governs how many concurrent calls can be
supported. Avaya 1100 Series IP Deskphones have the capability of handling 11 call
appearances, but display only one local call appearance button when idle (see display in
Section 7.4 Step 3), so the number of entries shown below are not required to match that
displayed on the telephone. Three are configured here to support conferencing scenarios.
add station 30043 Page 4 of 6
STATION
SITE DATA
Room: Headset? n
Jack: Speaker? n
Cable: Mounting: d
Floor: Cord Length: 0
Building: Set Color:
ABBREVIATED DIALING
List1: List2: List3:
BUTTON ASSIGNMENTS
1: call-appr 5:
2: call-appr 6:
3: call-appr 7:
4: 8:
5. Repeat Steps 1 - 4 as necessary to administer additional OPS stations and associations for
the SIP telephones.
5.11. Routing
Step Description
1. Enter the change aar analysis n command, where n is the number to be routed; in this
case 300 (matching any extensions starting with 300xx).
On Page 1 of the form configure the following fields as shown in the screen below:
Dialed String – Set to 300.
Total Min/Max – Set to 5
Route Patten - Set to the appropriate route pattern, in this case 60.
Call Type – Set to aar.
change aar analysis 3 Page 1 of 2
AAR DIGIT ANALYSIS TABLE
Location: all Percent Full: 0
BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR
0 1 2 M 4 W Request Dgts Format
Subaddress
1: y y y y y n n rest none
2: y y y y y n n rest none
3: y y y y y n n rest none
4: y y y y y n n rest none
5: y y y y y n n rest none
6: y y y y y n n rest none
Precedence Call Waiting? y
Session Manager is configured via System Manager. Use a web browser and enter “https://<ip-
address>/SMGR, where <ip-address> is the IP address of System Manager
Log in using the appropriate credentials. On the main configuration page, select User
Management under User Management, and click New to administer a new telephone user.
This will create a new User Profile. In the General section, enter a Last Name and First Name.
Note that fields market with * are required to be filled in.
In the Identity section, enter a Login Name, for example 30043@avaya.com, and the required
passwords, as shown on the next page. Note that the Shared Communication Profile password
is the one the telephone is required to use when registering to Session Manager. It is also
recommended to enter the display names. The Localized Display Name is what is displayed on a
telephone when a call is made.1 SMGR Login Password, while required, was not used in this
sample configuration, and can be any value.
1
When using Method 2 to configure telephone users (see Section 5.10), Session Manager uses this field to populate
the Name field in the station form in Communication Manager.
In the Communication Profile section, there are three sub-sections that need to be filled in:
Communication Address, Session Manager, and Station Profile. Clicking on the arrow next to
Communication Profile reveals the other sections.
Click on the box next to Session Manager, and select the appropriate Session Manager
Instance from the list. Select the appropriate Origination and Termination Application
Sequence.
The screen below shows what was used for extension 30043.
When done click at the bottom of the web page. Repeat the above steps for each
telephone to be configured.
2
This value for the Template applies for the 1120, 1140, and 1165 models.
Three models were tested: Avaya 1120E, 1140E, and 1165E. The configuration was done using
configuration files and the local telephone screen interface, as shown in these Application Notes.
The steps below show the configuration screens for the Avaya 1165E SIP Telephone.
Configuration files can be used for most options to support mass deployments.
The configuration steps are similar for all three telephones, the main difference being the number
of accounts or line appearances that each telephone supports. Make sure the number of lines used
matches what is configured in Communication Manager.
[DEVICE_CONFIG]
DOWNLOAD_MODE FORCED
VERSION 000100
PROTOCOL HTTP
FILENAME 1165DeviceConfig.dat
[FW]
DOWNLOAD_MODE FORCED
VERSION SIP1165e03.02.05.20
PROTOCOL HTTP
FILENAME SIP1165e03.02.05.20.bin
[DIALING_PLAN]
DOWNLOAD_MODE FORCED
VERSION 000020
PROTOCOL HTTP
#------DNS domain
DNS_DOMAIN ca.avaya.com
#------Server IP addresses
SERVER_IP1_1 10.1.1.4
SERVER_IP1_2 10.1.1.4
SERVER_IP3_1 47.103.241.74
SERVER_IP3_2 47.103.241.74
SERVER_IP4_1 47.11.43.24
SERVER_IP4_2 47.11.43.24
SERVER_IP5_1 47.11.33.25
SERVER_IP5_2 47.11.33.25
#------Listening ports
SIP_UDP_PORT 5060
SIP_TCP_PORT 5060
SIP_TLS_PORT 0
#------Server retries
SERVER_RETRIES1 3
SERVER_RETRIES2 3
SERVER_RETRIES3 3
#------Admin
ADMIN_PASSWORD 123456
ENABLE_LOCAL_ADMIN_UI YES
SECURE_UI_ENABLE NO
LOGOUT_WITHOUT_PASSWORD YES
SSH YES
SSHID 1234
SSHPWD 1234
#------Firmware update
AUTO_UPDATE YES
AUTO_UPDATE_TIME 0
#------Service Package
# Not supported in this configuration
ENABLE_SERVICE_PACKAGE NO
#------Banner
FORCE_BANNER YES
BANNER Avaya
#------Autologin
AUTOLOGIN_ENABLE YES
#------Time configuration
SNTP_ENABLE YES
SNTP_SERVER 10.1.1.21
TIMEZONE_OFFSET -18000
FORCE_TIME_ZONE No
#------VMAIL
VMAIL_DELAY 600
#------Expansion Module
EXP_MODULE_ENABLE YES
#------Mailbox entries
DEF_LANG English
MAX_INBOX_ENTRIES 100
MAX_OUTBOX_ENTRIES 100
MAX_REJECTREASONS 5
MAX_PRESENCENOTE 5
MAX_CALLSUBJECT 5
#------Bluetooth
ENABLE_BT YES
#------E911
E911_USERNAME 911
E911_PASSWORD 1234
E911_PROXY techtrial.com
E911_TXLOC INVITE
#------USB port
ENABLE_USB_PORT YES
USB_MOUSE UNLOCK
USB_KEYBOARD UNLOCK
USB_HEADSET UNLOCK
USB_MEMORY_STICK UNLOCK
#------Audio Codecs
AUDIO_CODEC1 G722
AUDIO_CODEC2 PCMU
AUDIO_CODEC3 G729
AUDIO_CODEC4 PCMA
AUDIO_CODEC5
AUDIO_CODEC6
AUDIO_CODEC7
AUDIO_CODEC8
G729_ENABLE_ANNEXB YES
# G723_ENABLE_ANNEXA YES
#------PROXY Checking
PROXY_CHECKING YES
#------File Manager
FM_CONFIG_ENABLE YES
FM_CERTS_ENABLE Y
#------DOD
DOD_ENABLE NO
#------DSCP Settings
DSCP_OAM 18
DSCP_CONTROL 40
DSCP_MEDIA_FLASHOVERRIDE 41
DSCP_MEDIA_FLASH 42
DSCP_MEDIA_IMMEDIATE 44
DSCP_MEDIA_PRIORITY 45
DSCP_MEDIA 46
#------Login banner
LOGIN_BANNER_ENABLE NO
#------NAT signaling
NAT_SIGNALLING SIP_PING
BG_IMAGE_ENABLE YES
BG_IMG_SELECT_ENABLE YES
USE_BG_IMAGE screensaver2.jpg
#------Fonts
OUTLINEFONT_ENABLE YES
FONTSMOOTH_ENABLE YES
#------BLF
BLF_ENABLE No
#------End
/* ------------------------------------------------------------------- */
/* */
/* Avaya 1100-series IP Deskphone Dial Plan */
/* */
/* ------------------------------------------------------------------- */
/* Domain used in the dialed URL of the SIP INVITE message */
$n="avaya.com"
$t=300
%%
7.4. Configure Speed Dial Buttons for Avaya Extended Feature Set
Additional Communication Manager features can be accessed by dialing the corresponding FNE.
For example, if the telephone has been defined in Communication Manager as part of a pickup
group, then dial the Call Pickup FNE (in this case 70010) to answer a call to any member of that
group. Features that involve an existing call (e.g., Call Park) will require putting that call on
hold, and placing a new call using the appropriate FNE. Holding the existing call is done
automatically by the telephone if another call is placed. This procedure can be streamlined by
using free line appearance buttons on the telephone for speed dialing. Commonly used FNEs can
be defined on these buttons, in many cases facilitating one-button feature access.
The following steps describe how to configure Avaya 1100-Series IP Deskphones with speed
dial buttons. This technique is most useful with models that have many line appearance buttons,
such as the 1140E and 1165E. Section 7.4.1 shows how to manually configure speed dial
buttons at each individual phone. For mass deployments, Section 7.4.2 shows how the device
configuration file and a speed dial list file can be used to support automatic configuration. Note
that manually configured buttons will override automatically configured buttons at the same
position. See Reference [8] for more details.
Left Right
8 4
7 3
6 2
5 1
DEFAULT_CUSTOMKEYSFILE SpeedDials.txt
2. Create the file SpeedDials.txt with an entry for each speed dial button that is to be programmed.
Set index to the key position number (see layout for the 1165E in Step 1 in Section 7.4.1), label
to the desired text to be displayed at the button position, target to FNE@domain, where FNE is
the extension of the FNE (see Section 5.8 Step 3), and domain is the domain configured in
Session Manager. The example below corresponds to the Pickup button configured for the
1165E, as displayed in Step 3 of Section 7.4.1).
[key]
index=2
label=Pickup
target=70010@avaya.com
type=spdial
3. Reboot the phone, and it will automatically program the specified speed dial buttons.
Step Description
1. After rebooting the telephone, use the More and Prefs soft keys at the phone to verify that
the parameters set in the phone configuration file have been loaded. Verify registration
with Session Manager by the appearance of the idle screen. If this is the first time
registration is being attempted and multiple domains have been configured, enter the
appropriate domain (“avaya.com” in the sample configuration). Verify that the line
appearance shows the Communication Manager extension for that phone.
2. Verify basic feature set administration by lifting the handset (or pressing the speaker
button), and making calls to other phones. Test supported features according to Table 2
and feature deployment plans at the site.
3. Enter the status trunk n command, where n is the SIP trunk configured in Section 5.6.
Note down the Member with Service State set to in-service/active. In this example,
0060/006 and 0060/007 are active and either member can be used to verify whether calls
shuffled and which codec was used.
status trunk 60 Page 1
Video Near:
Video Far:
Video Port:
Video Near-end Codec: Video Far-end Codec:
5. Verify that speed dial buttons defined locally at the phone are displayed. If any are
missing or are inoperative, check the local settings or the configuration file.
6. Verify additional Communication Manager features by pressing the speed dial button for
the feature, or lifting the handset and dialing the FNE. If busy or intercept tone is heard,
check Communication Manager for the correct FNE, proper permissions under
COS/COR, and the proper station button assignment to support the feature.
7. Call a telephone that currently has no voice messages, and leave a message. Verify that
the message-waiting indicator illuminates on the called telephone. Press the messages
button on that telephone and verify that the voice messaging system is called. Use the
voice messaging menus to retrieve and delete the voice message, verifying that DTMF is
interpreted correctly by the system, and that the message waiting indicator extinguishes.
9. Conclusion
These Application Notes have described the administration steps required to use Avaya 1100-
Series IP Deskphones with SIP software with Session Manager, Communication Manager, and
Modular Messaging. Basic, supplementary, and extended feature sets were covered. The
extended set relies on Communication Manager Feature Server and Feature Name Extensions to
support additional SIPPING features described in RFC 5359.
[1] Administering Avaya Aura™ Communication Manager, Release 5.2, Issue 5.0, May 2009,
Document Number 03-300509.
[2] Administering Network Connectivity on Avaya Aura™ Communication Manager, Issue 14,
May 2009, Document Number 555-233-504.
[3] SIP Support in Avaya Aura™ Communication Manager Running on Avaya S8xxx Servers,
Issue 9, May 2009, Document Number 555-245-206.
[4] Administering Avaya Aura™ Session Manager, Release 5.2, Issue 2.0, November 2009,
Document Number 03-603324.
[5] Avaya Aura™ Communication Manager Screen Reference, Issue 1.0, May 2009, Document
Number 03-602878.
[6] Administering Avaya Aura™ Communication Manager as a Feature Server, Release 5.2,
Issue 1.2, January 2010, Document Number 03-603479.
[7] Configuring 9600-Series SIP Phones with Avaya AuraTM Session Manager Release 5.2 –
Issue 1.0, February 2010, Avaya Solution Interoperability Lab Application Notes.
[9] Session Initiation Protocol Service Examples, Internet Engineering Task Force, RFC 5259,
available at http://www.ietf.org.
[10] Modular Messaging Release 5.2 with Avaya MSS, Messaging Application Server (MAS)
Administration Guide, November 2009.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya Solution &
Interoperability Test Lab at interoplabnotes@list.avaya.com.